PsiMedia

Hello,

What's the status on PsiMedia? Is it still being worked on?
(last  
commit was two months ago)

BTW, why GStreamer instead of ffmpeg?

-- 
Pau Garcia i Quiles
http://www.elpauer.org
(Due to my workload, I may need 10 days to answer)

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Hi,

On Thursday 11 September 2008 04:50:44 Pau Garcia i Quiles
wrote:
> What's the status on PsiMedia? Is it still being worked
on? (last
> commit was two months ago)

It's not currently being worked on.  The plan is to return
to it around 
November.  At the XMPP Summit I was convinced by Robert
McQueen to use 
Farsight (a library that augments GStreamer for the kind of
stuff we're 
doing), and so that'll probably be one of the first items on
the agenda.

> BTW, why GStreamer instead of ffmpeg?

I did not look at ffmpeg specifically, but GStreamer has
ffmpeg-based plugins 
for handling various codec types (perhaps ffmpeg handles
more than codecs, in 
that case I admit I'm not aware of it).

GStreamer handles audio/video hardware, codecs, and RTP
payloading, which is 
everything we want and need.

-Justin
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On Monday 10 November 2008 21:25:51 Justin Karneges wrote:
> On Thursday 06 November 2008 14:00:53 Justin Karneges
wrote:
> > I still need to rewire the implementation and
example application before
> > I commit these changes.  I'll post again when it
is done.
>
> Done. [...]

Regarding that, I've started refactoring the implementation
this week.  I even 
did some profiling.  The profiling was pure curiosity, and I
discovered that 
waking the glib thread every 20ms for Speex packets was too
resource 
intensive.  Also, it seems that audio resampling and Speex
encoding are not 
cheap operations.  See this bug entry:
http
://bugzilla.gnome.org/show_bug.cgi?id=385061

I was doing tests using the file mode of PsiMedia.  My test
file is 30fps 
480x272 Theora, 44.1KHz Vorbis (which then gets transcoded
to Theora 
(redundantly) and 16KHz Speex, for transmission).  The 20ms
wake combined 
with the high cost of audio transcoding resulted in the app
tripping over 
itself.  It would slowly creep to 100% cpu at which point
performance would 
then start degrading.  When I smoothed out the glib waking
so that it didn't 
occur more frequently than, say, 40ms, then I got acceptable
performance.  
However, even in that case, the cpu usage of my 1.8GHz AMD
would hover 
between 70-80%.

I don't think there's a whole lot more I can do.  The rest
is mostly a matter 
of further GStreamer optimization (such as the above bug
entry), and of 
course avoiding codec configurations where transcoding costs
become 
overbearing.  80% cpu is a lot, but the above scenario is
not realistic for a 
voip session.  I'll try to get some benchmarks for live
transmission, since 
in real life you won't be doing vorbis->speex on the
fly.

Other note of interest: It seems that if I wake the Qt
thread instead of the 
glib thread for processing GStreamer events, then the 20ms
Speex wake 
produces acceptable results.  I don't know if this means
the Qt event loop is 
more optimized than the glib event loop, or if maybe Qt does
its own 
smoothing of the events (or maybe even something else is
going on...).  In 
any case, I'll go with waking the Qt thread as this cuts
out the middle man 
and ultimately reduces latency anyway.  Instead of RTP
packets travelling 
GStreamer -> glib -> Qt, it just goes GStreamer ->
Qt.

Another note: I still haven't written any Farsight-specific
code.  I think for 
the first iteration, I'm going to leave Farsight out.  This
will mean fixed 
codecs and no RTCP.  A later development cycle will bring in
Farsight to tie 
up the loose ends.

I think I have about a week left of refactoring and then we
can start having 
users test the demo app with seriousness.

-Justin
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